Cisco Unified SCCP and SIP SRST System Administrator Guide (All Versions)
Integrating Voice Mail with Cisco Unified SRST
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Table of Contents

Integrating Voice Mail with Cisco Unified SRST

Contents

Information About Integrating Voice Mail with Cisco Unified SCCP SRST

How to Integrate Voice Mail with Cisco Unified SCCP and SIP SRST

Configuring Direct Access to Voice Mail

Examples

Configuring Message Buttons

Examples

Redirecting to Cisco Unified Communications Manager Gateway

Configuring Call Forwarding to Voice Mail

Call Routing Instructions Using DTMF Digit Patterns

Prerequisites

Examples

Configuring Message Waiting Indication (Cisco Unified SCCP SRST Routers)

Configuring Message Waiting Indication (SIP Phones in SRST Mode)

Configuration Examples for SCCP SRST

Configuring Local Voice-Mail System (FXO and FXS): Example

Configuring Central Location Voice-Mail System (FXO and FXS): Example

Configuring Voice-Mail Access over FXO and FXS: Example

Configuring Voice-Mail Access over BRI and PRI: Example

Message Waiting Indication for SIP SRST: Example

How to Configure DTMF Relay for SIP Applications and Voice Mail

DTMF Relay Using SIP RFC2833

Troubleshooting Tips

DTMF Relay Using SIP Notify (Nonstandard)

Troubleshooting Tips

Where to Go Next

Integrating Voice Mail with Cisco Unified SRST

This chapter describes how to make your existing voice-mail system run on phones connected to a Cisco Unified SRST router during Cisco Unified Communications Manager fallback.

Cisco Unified SRST also supports incoming and outgoing Session Initiation Protocol (SIP) calls to and from Cisco Unified IP phones and router voice gateway voice ports. SIP may be used in situations where the Cisco Unified SRST Router is separate from the PSTN gateway and the SRST and PSTN gateways are linked together using SIP (instead of H.323).

For more information about SIP, see Cisco IOS SIP Configuration Guide.

Information About Integrating Voice Mail with
Cisco Unified SCCP SRST

Cisco Unified SCCP SRST can send and receive voice-mail messages from Cisco Unity and other voice-mail systems during Cisco Unified CM fallback. When the WAN is down, a voice-mail system with BRI or PRI access to the Cisco Unified SCCP SRST system uses ISDN signaling (see Figure 10-1). Systems with Foreign Exchange Office (FXO) or Foreign Exchange Station (FXS) access connect to a PSTN and use in-band dual tone multifrequency (DTMF) signaling (see Figure 10-2).

Figure 10-1 Cisco Unified Communications Manager Fallback with BRI or PRI

 

Figure 10-2 Cisco Unified Communications Manager Fallback with PSTN

 

Both configurations allow phone message buttons to remain active and calls to busy or unanswered numbers to be forwarded to the dialed numbers’ mailboxes.

Calls that reach a busy signal, calls that are unanswered, and calls made by pressing the message button are forwarded to the voice-mail system. To make this happen, you must configure access from the dial peers to the voice-mail system and establish routing to the voice-mail system for busy and unanswered calls and for message buttons.

If the voice-mail system is accessed over FXO or FXS, you must configure instructions (DTMF patterns) for the voice-mail system so that it can access the correct voice-mail system mailbox. If your voice-mail system is accessed over BRI or PRI, no instructions are necessary because the voice-mail system can log in to the calling phone’s mailbox directly.

How to Integrate Voice Mail with Cisco Unified SCCP and SIP SRST

This section contains the following tasks:


NoteSupport for SIP SRST is added from IOS release 15.1(4)M3 and 15.2(1)T2. Support for SIP SRST is added from IOS release 15.1(4)M3 and 15.2(1)T2.


Configuring Direct Access to Voice Mail

To access voice-mail messages with FXO or FXS access, you must have POTS dial peers configured with a destination pattern that matches the voice-mail system’s number. Also, you must associate the dial peer with the port to which the voice-mail system is accessed.

Both sets of configurations are done in global configuration mode and in dial-peer configuration mode. The summary and detailed steps below include only the basic commands necessary to perform this task. You may require additional commands for your particular dial-peer configuration.

SUMMARY STEPS

1. dial-peer voice tag { pots | voatm | vofr | voip }

2. destination-pattern [ + ] string [ T ]

3. port { slot-number / subunit-number / port | slot / port : ds0-group-no }

4. forward-digits { num-digit | all | extra }

5. exit

DETAILED STEPS

 

Command or Action
Purpose

Step 1

dial-peer voice tag { pots | voatm | vofr | voip }

 
Router(config)# dial-peer voice 1002 pots

(FXO or FXS and BRI or PRI) Defines a particular dial peer, specifies the method of voice encapsulation, and enters dial-peer configuration mode. The dial-peer command provides different syntax for individual routers. This example is syntax for Cisco 3600 series routers.

  • tag : Digits that define a particular dial peer. Range is from 1 to 2147483647.
  • pots : Indicates that this is a POTS dial peer that uses VoIP encapsulation on the IP backbone.
  • voatm : Specifies that this is a VoATM dial peer that uses real-time AAL5 voice encapsulation on the ATM backbone network.
  • vofr : Specifies that this is a VoFR dial peer that uses FRF.11 encapsulation on the Frame Relay backbone network.
  • voip : Indicates that this is a VoIP dial peer that uses voice encapsulation on the POTS network.

Step 2

destination-pattern [ + ] string [ T ]
 
Router(config-dial-peer)# destination-pattern 1100T
 

(FXO or FXS and BRI or PRI) Specifies either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer.

  • + : (Optional) Character that indicates an E.164 standard number.
  • string : See Table 10-1 .
  • T : (Optional) Control character that indicates that the destination-pattern value is a variable-length dial string.

Step 3

port { slot-number / subunit-number / port | slot / port : ds0-group-no }
 
Router(config-dial-peer)# port 1/1/1
 

(FXO or FXS and BRI or PRI) Associates a dial peer with a specific voice port on Cisco 3600 series routers.

  • slot-number : Number of the slot in the router in which the voice interface card (VIC) is installed. Valid entries are from 0 to 3, depending on the slot in which it is installed.
  • subunit-number : Subunit on the VIC in which the voice port is located. Valid entries are 0 or 1.
  • port : Voice port number. Valid entries are 0 and 1.
  • ds0-group-no : Specifies the DS0 group number. Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card.

Step 4

forward-digits { num-digit | all | extra }

 
Router(config-dial-peer)# forward-digits all

(Optional for FXO or FXS) Specifies which digits to forward for voice calls.

  • num-digit : The number of digits to be forwarded. If the number of digits is greater than the length of a destination phone number, the length of the destination number is used. Range is 0 to 32. Setting the value to 0 is equivalent to entering the no forward-digits command.
  • all : Forwards all digits. If all is entered, the full length of the destination pattern is used.
  • extra : If the length of the dialed digit string is greater than the length of the dial-peer destination pattern, the extra right-justified digits are forwarded. However, if the dial-peer destination pattern is variable length and ends with the character “T” (for example: T, 123T, 123...T), extra digits are not forwarded.

Step 5

exit

 
Router(config-dial-peer)# exit

(FXO or FXS and BRI or PRI) Exits dial-peer configuration mode.

 

Table 10-1 Valid Entries for the String Argument in the destination-pattern command

Entry
Description

Digits 0 to 9

Letters A through D

Asterisk (*) and pound sign (#)

These appear on standard touch-tone dial pads.

Comma (,)

Inserts a pause between digits.

Period (.)

Matches any entered digit (this character is used as a wildcard).

Percent sign (%)

Indicates that the preceding digit occurred zero or more times; similar to the wildcard usage.

Plus sign (+)

Indicates that the preceding digit occurred one or more times.

Note The plus sign used as part of a digit string is different from the plus sign that can be used in front of a digit string to indicate that the string is an E.164 standard number.

Circumflex (^)

Indicates a match to the beginning of the string.

Parentheses ( ( ) ), which indicate a pattern and are the same as the regular expression rule.

Dollar sign ($)

Matches the null string at the end of the input string.

Backslash symbol (\)

Is followed by a single character and matches that character. Can be used with a single character with no other significance (matching that character).

Question mark (?)

Indicates that the preceding digit occurred zero or one time.

Brackets ( [ ] )

Indicates a range. A range is a sequence of characters enclosed in the brackets; only numeric characters from 0 to 9 are allowed in the range.

Examples

The following FXO and FXS example sets up a POTS dial peer named 1102, matches dial-peer 1102 to voice-mail extension 1101, and assigns dial-peer 1102 to voice-port 1/1/1 where the voice-mail system is connected. Other dial peers are configured for direct access to voice mail.

voice-port 1/1/1
timing digit 250
timing inter-digit 250
 
dial-peer voice 1102 pots
destination-pattern 1101
port 1/1/1
forward-digits all
 
dial-peer voice 1103 pots
destination-pattern 1101
port 1/1/1
forward-digits all
 
dial-peer voice 1104 pots
destination-pattern 1101
port 1/1/1
forward-digits all
 

The following example sets up a POTS dial peer named 1102 to go directly to 1101 through port 2/0:23:

controller T1 2/0
framing esf
clock source line primary
linecode b8zs
cablelength short 133
pri-group timeslots 21-24
 
interface Serial2/0:23
no ip address
no logging event link-status
isdn switch-type primary-net5
isdn incoming-voice voice
isdn T309-enable
no cdp enable
 
voice-port 2/0:23
 
dial-peer voice 1102 pots
destination-pattern 1101T
port 2/0:23

Configuring Message Buttons

To activate the message buttons on Cisco Unified IP phones connected to the Cisco Unified SCCP and SIP SRST router during Cisco Unified Communications Manager fallback, you must program a speed-dial number to the voice-mail system. The speed-dial number is dialed when message buttons on phones connected to the Cisco Unified SCCP and SIP SRST router are pressed during Cisco Unified CM fallback. In addition, call forwarding must be configured so that calls to busy and unanswered numbers are sent to the voice-mail number.

This configuration is required for FXO or FXS and BRI or PRI.

SUMMARY STEPS

1. call-manager-fallback

2. voicemail phone-number

3. call-forward busy directory-number

4. call-forward noan directory-number timeout seconds

5. exit

6. voice register pool tag

7. call-forward b2bua busy directory-number

8. call-forward b2bua noan directory-number timeout seconds

9. exit

DETAILED STEPS

 

Command or Action
Purpose

Step 1

call-manager-fallback

 

Router(config)# call-manager-fallback

Enters call-manager-fallback configuration mode.

Step 2

voicemail phone-number

 
Router(config-cm-fallback)# voicemail 5550100

Configures the telephone number that is dialed when the message button on a Cisco Unified SCCP IP Phone is pressed.

  • phone-number : Phone number configured as a speed-dial number for retrieving messages.

Step 3

call-forward busy directory-number
 
Router(config-cm-fallback)# call-forward busy 2000

Configures call forwarding to another number when the Cisco SCCP IP phone is busy.

  • directory-number : Selected directory number representing a fully qualified E.164 number. This number can contain “.” wildcard characters that correspond to the right-justified digits in the directory number extension.

Step 4

call-forward noan directory-number timeout seconds
 
Router(config-cm-fallback)# call-forward noan 2000 timeout 10

Configures call forwarding to another number when no answer is received from the Cisco SCCP IP phone.

  • directory-number : Selected directory number representing a fully qualified E.164 number. This number can contain “.” wildcard characters that correspond to the right-justified digits in the directory number extension.
  • timeout seconds : Sets the waiting time, in seconds, before the call is forwarded to another phone. The seconds range is from 3 to 60000.

Step 5

exit

 

Router(config-cm-fallback)# exit

Exits call-manager-fallback configuration mode.

Step 6

voice register pool tag

 

Router(config)# voice register pool 1

Enters voice register pool configuration mode.

Step 7

call-forward b2bua busy directory-number

 

Router(config-register-pool)# call-forward b2bua busy 2000

Configures call forwarding to another number when the Cisco SIP IP phone is busy.

  • directory-number : Selected directory number representing a fully qualified E.164 number. This number can contain “.” wildcard characters that correspond to the right-justified digits in the directory number extension.

Step 8

call-forward b2bua noan directory-number timeout seconds

 

Router(config-register-pool)# call-forward noan 2000 timeout 10

Configures call forwarding to another number when no answer is received from the Cisco SIP IP phone.

  • directory-number : Selected directory number representing a fully qualified E.164 number. This number can contain “.” wildcard characters that correspond to the right-justified digits in the directory number extension.
  • timeout seconds : Sets the waiting time, in seconds, before the call is forwarded to another phone. The seconds range is from 3 to 60000.

Step 9

exit

 

Router(config-register-pool)# exit

Exits voice register pool configuration mode.

Examples

The following example specifies 1101 as the speed-dial number that is issued when message buttons are pressed on Cisco Unified IP Phones connected to the Cisco Unified SRST router. All busy and unanswered calls are configured to be forwarded to the voice-mail number (1101).

call-manager-fallback
voicemail 1101
call-forward busy 1101
call-forward noan 1101 timeout 3
voice register pool 1
call-forward b2bua busy 1101
call-forward b2bua noan 1101 timeout 3
 

Redirecting to Cisco Unified Communications Manager Gateway


NoteThe following task is required for voice-mail systems with BRI or PRI access. The following task is required for voice-mail systems with BRI or PRI access.


In addition to supporting message buttons for retrieving personal messages, Cisco Unified SCCP SRST allows the automatic forwarding of calls to busy and unanswered numbers to voice-mail systems. Voice-mail systems with BRI or PRI access can log in to the calling phone’s mailbox directly. For this to happen, some Cisco Unified CM configuration is recommended. If your voice-mail system supports Redirected Dialed Number Identification Service (RDNIS), RDNIS must be included in the outgoing SETUP message to Cisco Unified CM to declare the last redirected number and the originally dialed number to and from configured devices and applications.


Step 1 From any page in Cisco Unified CM, click Device and Gateway .

Step 2 From the Find and List Gateways page, click Find.

Step 3 From the Find and List Gateways page, choose a device name.

Step 4 From the Gateway Configuration page, check Redirecting Number IE Delivery - Outgoing .

Configuring Call Forwarding to Voice Mail


NoteThe following task is required for voice-mail systems with FXO or FXS access. The following task is required for voice-mail systems with FXO or FXS access.


In addition to supporting message buttons for retrieving personal messages, Cisco Unified SCCP SRST allows the automatic forwarding of calls to busy or unanswered numbers to voice-mail systems. The forwarded calls can be routed to almost any location in the voice-mail system. Typically, calls are forwarded to a location in the called number’s mailbox where the caller can leave messages.

Call Routing Instructions Using DTMF Digit Patterns

Cisco Unified SCCP SRST call-routing instructions are required so that forwarded calls can be sent to the correct voice mailboxes. These instructions consist of DTMF digits configured in patterns that match the dial sequences required by the voice-mail system to get to a particular voice-mail location. For example, a voice-mail system may be designed so that callers must do the following to leave a message:

1. Dial the central voice-mail number (1101) and press #.

2. Dial an extension number (6000) and press #.

3. Dial 2 to select the menu option for leaving messages in the extension number’s mailbox.

For Cisco Unified SCCP SRST to forward a call to a busy or unanswered number to extension 6000’s mailbox, it must be programmed to issue a sequence of 1101#6000#2. As shown in Figure 10-3, this is accomplished through the voicemail and pattern commands.

Figure 10-3 How Voice-Mail Dial Sequence 1101#6000#2 Is Configured in Cisco Unified SCCP SRST

 

The # cgn #2, # cdn #2, and # fdn #2 portions of the pattern commands shown in Figure 10-3 are DTMF digit patterns. These patterns are composed of tags and tokens. Tags are sets of characters representing DTMF tones. Tokens consist of three command keywords ( cgn , cdn , and fdn ) that declare the state of an incoming call transferred to voice mail.

A tag can be up to three character from the DTMF tone set (A to D, 0 to 9, # and *). Voice-mail systems can use limited sets of DTMF tones. For example, Cisco Unity uses all DTMF tones but A to D. Tones can be defined in multiple ways. For example, when the star (*) is placed in front of a token by itself, it can mean “dial the following token number,” or, if it is at the end of a token, it can mark the end of a token number. If the asterisk is between other tag characters, it can mean dial *. The use of tags depends on how DTMF tones are defined by your voice-mail system.

Tokens tell Cisco Unified SSCCP RST what telephone number in the call forwarding chain to use in the pattern. As shown in Figure 10-4, there are three types of tokens that correspond to three possible call states during voice-mail forwarding.

Figure 10-4 How Numbers Are Extracted from Tokens

 

Sets of tags and tokens or patterns activate a voice-mail system when one ofthe following occurs:

  • A user presses the message button on a phone ( pattern direct command).
  • An internal extension attempts to connect to a busy extension and the call is forwarded to voice mail ( pattern ext-to-ext busy command).
  • An internal extension fails to connect to an extension and the call is forwarded to voice mail ( pattern ext-to-ext no-answer command).
  • An external trunk call reaches a busy extension and the call is forwarded to voice mail ( pattern trunk-to-ext busy command).
  • An external trunk call reaches an unanswered extension and the call is forwarded to voice mail ( pattern trunk-to-ext no-answer command).

Prerequisites

You can find information about how Cisco Unity handles voice-mail calls in the How to Transfer a Caller Directly into a Cisco Unity Mailbox document. Additional call-handling information can be found in the “Subscriber and Operator Orientation” chapters of any Cisco Unity system administration guide.

For other voice-mail systems, see the analog voice mail integration configuration guide or information about the system’s call handling.

SUMMARY STEPS

1. vm-integration

2. pattern direct tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }]
[
tag3 { CGN | CDN | FDN }] [ last-tag ]

3. pattern ext-to-ext busy tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }]
[
tag3 { CGN | CDN | FDN }] [ last-tag ]

4. pattern ext-to-ext no-answer tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }]
[
tag3 { CGN | CDN | FDN }] [ last-tag ]

5. pattern trunk-to-ext busy tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }]
[
tag3 { CGN | CDN | FDN }] [ last-tag ]

6. pattern trunk-to-ext no-answer tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }]
[
tag3 { CGN | CDN | FDN }] [ last-tag ]

DETAILED STEPS

 

Command or Action
Purpose

Step 1

vm-integration

 
Router(config)# vm-integration

Enters voice-mail integration mode and enables voice-mail integration with DTMF and analog voice-mail systems.

Step 2

pattern direct tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }] [ tag3 { CGN | CDN | FDN }] [ last-tag ]

 

Router(config-vm-int)# pattern direct 2 CGN *

Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when the user presses the messages button on the phone.

  • tag1 : Alphanumeric string fewer than four DTMF digits in length. The alphanumeric string consists of a combination of four letters (A, B, C, and D), two symbols (* and #), and ten digits (0 to 9). The tag numbers match the numbers defined in the voice-mail system’s integration file, immediately preceding either the number of the calling party, the number of the called party, or a forwarding number.
  • tag2 and tag3 : (Optional) See tag1 .
  • last-tag : See tag1 . This tag indicates the end of the pattern.
  • CGN : Calling number (CGN) information is sent to the voice-mail system.
  • CDN : Called number (CDN) information is sent to the voice-mail system.
  • FDN : Forwarding number (FDN) information is sent to the voice-mail system.

Step 3

pattern ext-to-ext busy tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }] [ tag3 { CGN | CDN | FDN }] [ last-tag ]

 

Router(config-vm-int)# pattern ext-to-ext busy 7 FDN * CGN *

Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension attempts to connect to a busy extension and the call is forwarded to voice mail. For argument and keyword information, see .

Step 4

pattern ext-to-ext no-answer tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }] [ tag3 { CGN | CDN | FDN }] [ last-tag ]

 

Router(config-vm-int)# pattern ext-to-ext no-answer 5 FDN * CGN *

Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension fails to connect to an extension and the call is forwarded to voice mail. For argument and keyword information, see .

Step 5

pattern trunk-to-ext busy tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }] [ tag3 { CGN | CDN | FDN }] [ last-tag ]

 

Router(config-vm-int)# pattern trunk-to-ext busy 6 FDN * CGN *

Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an external trunk call reaches a busy extension and the call is forwarded to voice mail. For argument and keyword information, see .

Step 6

pattern trunk-to-ext no-answer tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }] [ tag3 { CGN | CDN | FDN }] [ last-tag ]

 

Router(config-vm-int)# pattern trunk-to-ext no-answer 4 FDN * CGN *

Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an external trunk call reaches an unanswered extension and the call is forwarded to voice mail. For argument and keyword information, see .

Examples

For the following configuration, if the voice-mail number is 1101, and 3001 is a phone with a message button, 1101*3001 would be dialed automatically when the 3001 message button is pressed. Under these circumstances, 3001 is considered to be a calling number or inbound call number.

vm-integration
pattern direct * CGN
 

For the following configuration, if 3001 calls 3006 and 3006 does not answer, the SCCP SRST router will forward 3001 to the voice-mail system (1101) and send to the voice-mail system the DTMF pattern # 3006 #2. This pattern is intended to select voice mailbox number 3006 (3006’s voice mailbox). For this pattern to be sent, 3001 must be a forwarding number.

vm-integration
pattern ext-to-ext no-answer # FDN #2
 

For the following configuration, if 3006 is busy and 3001 calls 3006, the SCCP SRST router will forward 3001 to the voice-mail system (1101) and send to the voice-mail system the DTMF pattern # 3006 #2. This pattern is intended to select voice mailbox number 3006 (3006’s voice mailbox). For this pattern to be sent, 3001 must be a forwarding number.

vm-integration
pattern ext-to-ext busy # FDN #2

Configuring Message Waiting Indication (Cisco Unified SCCP SRST Routers)

The MWI relay mechanism is initiated after someone leaves a voice-mail message on the remote voice-mail message system. MWI relay is required when one Cisco Unity Voice Mail system is shared by multiple Cisco Unified SCCP SRST routers. SCCP SRST routers use the SIP Subscribe and Notify methods for MWI. See Configuring Cisco IOS SIP Configuration Guide for more information on SIP MWI and the Subscribe and Notify methods. The SCCP SRST router that is the SIP MWI relay server acts as the SIP notifier. The other remote routers act as the SIP subscribers.

SUMMARY STEPS

1. call-manager-fallback

2. mwi relay

3. mwi reg-e164

4. exit

5. sip-ua

6. mwi-server { ipv4: destination-address | dns: host-name } [ expires seconds ] [port port ]
[ transport { tcp | udp }] [ unsolicited ]

7. exit

DETAILED STEPS

 

Command
Purpose

Step 1

call-manager-fallback

 

Router(config)# call-manager-fallback

Enters call-manager-fallback configuration mode.

Step 2

mwi relay

 

Router(config-cm-fallback)# mwi relay

Enables the SCCP SRST router to relay MWI information to remote Cisco IP phones.

Step 3

mwi reg-e164

 

Router(config-cm-fallback)# mwi reg-e164

Registers E.164 numbers rather than extension numbers with a SIP proxy or registrar.

Step 4

exit

 

Router(config-cm-fallback)# exit

Exits call-manager-fallback configuration mode.

Step 5

sip-ua

 

Router(config)# sip-ua

Enters SIP user-agent configuration mode.

Step 6

mwi-server { ipv4: destination - address | dns : host - name } [ expires seconds ] [ port port ]
[ transport { tcp | udp }] [ unsolicited ]

 

Router(config-sip-ua)# mwi-server ipv4:10.0.2.254

 

Configures voice-mail server settings on a voice gateway or user agent. The IP address and port for the SIP-based MWI server should be in the same LAN as the voice-mail server. The MWI server is a Cisco Unified SCCP SRST router. Keywords and arguments are as follows:

  • ipv4: destination-address : IP address of the voice-mail server.
  • dns: host-name : The argument should contain the complete hostname to be associated with the target address; for example, dns:test.cisco.com .
  • expires seconds : Subscription expiration time, in seconds. Range is from 1 to 999999. Default is 3600.
  • port port : Port number on the voice-mail server. Default is 5060.
  • transport : Transport protocol to the voice-mail server. Valid values are tcp and udp. Default is UDP.
  • unsolicited : Requires the voice-mail server to send a SIP notification message to the voice gateway or UA if the mailbox status changes. Removes the requirement that the voice gateway subscribe for MWI service.

Step 7

exit

 

Router(config-sip-ua)# exit

Exits SIP user-agent configuration mode.

Configuring Message Waiting Indication (SIP Phones in SRST Mode)

On SIP phones operating in the SIP SRST mode, you can use the mwi unsolicited command to configure a message-waiting notification when a message is sent by the Cisco Unity Express (CUE). The SIP phone then displays the notification when indicated by the voice messaging system. To configure message-waiting notification, perform the following steps.

SUMMARY STEPS

1. enable

2. configure terminal

3. sip-ua

4. mwi-server { ipv4: destination-address | dns: host-name } [ unsolicited ]

5. exit

6. voice register global

7. mwi unsolicited

8. end

DETAILED STEPS

 

Command
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

sip-ua

 

Router(config)# sip-ua

Enters Session Initiation Protocol (SIP) user agent (ua) configuration mode for configuring the user agent.

Step 4

mwi-server { ipv4: destination - address | dns : host - name } [ unsolicited ]

 

Router(config-sip-ua)# mwi-server ipv4:10.0.2.254 unsolicited

 

Or

Router(config-sip-ua)# mwi-server dns:server.yourcompany.com unsolicited

Configures voice-mail server settings on a voice gateway or user agent. Keywords and arguments are as follows:

  • ipv4: destination-address : IP address of the voice-mail server.
  • dns: host-name : The argument should contain the complete hostname to be associated with the target address; for example, dns:test.cisco.com .
  • unsolicited : Requires the voice-mail server to send a SIP notification message to the voice gateway or UA if the mailbox status changes. Removes the requirement that the voice gateway subscribe for MWI service.

Step 5

exit

 

Router(config-sip-ua)# exit

Exits SIP user-agent configuration mode.

Step 6

voice register global

 

Router(config)# voice register global

Enters voice register global configuration mode to set parameters for all supported SIP phones in SIP SRST mode.

Step 7

mwi unsolicited

 

Router(config-register-global)# mwi unsolicited

Enables all SIP phones to receive MWI notification.

Step 8

end

 

Router(config-register-global)# end

Exits to privileged EXEC mode.

Configuration Examples for SCCP SRST

This section provides the following configuration examples:

Configuring Local Voice-Mail System (FXO and FXS): Example

The “Dial-Peer Configuration for Integration of Voice-Mail with Cisco Unified SCCP SRST” section of the example below shows a legacy dial-peer configuration for a local voice-mail system. The “Cisco Unified SCCP SRST Voice-Mail Integration Pattern Configuration” section must be compatible with your voice-mail system configuration.

! Dial-Peer Configuration for Integration of Voice-Mail with Cisco Unified SRST
!
dial-peer voice 101 pots
destination-pattern 14011
port 3/0/0
!
dial-peer voice 102 pots
preference 1
destination-pattern 14011
port 3/0/1
!
dial-peer voice 103 pots
preference 2
destination-pattern 14011
port 3/1/0
!
dial-peer voice 104 pots
destination-pattern 14011
port 3/1/1
!
! Cisco Unified SRST configuration
!
call-manager-fallback
max-ephones 24
max-dn 144
ip source-address 1.4.214.104 port 2000
voicemail 14011
call-forward busy 14011
call-forward noan 14011 timeout 3
 
! Cisco Unified SRST Voice-Mail Integration Pattern Configuration
!
vm-integration
pattern direct 2 CGN *
pattern ext-to-ext no-answer 5 FDN * CGN *
pattern ext-to-ext busy 7 FDN * CGN *
pattern trunk-to-ext no-answer 4 FDN * CGN *
pattern trunk-to-ext busy 6 FDN * CGN *

Configuring Central Location Voice-Mail System (FXO and FXS): Example

The “Dial-Peer Configuration for Integration of Voice-Mail with Cisco Unified SCCP SRST in Central Location” section of the example shows a legacy dial-peer configuration for a central voice-mail system. The “Cisco Unified SCCP SRST Voice-Mail Integration Pattern Configuration” section must be compatible with your voice-mail system configuration.


NoteMessage waiting indicator (MWI) integration is not supported for PSTN access to voice-mail systems at central locations. Message waiting indicator (MWI) integration is not supported for PSTN access to voice-mail systems at central locations.


! Dial-Peer Configuration for Integration of Voice-Mail with Cisco Unified SRST in Central
! Location
!
dial-peer voice 101 pots
destination-pattern 14011
port 3/0/0
!
! Cisco Unified SRST configuration
!
call-manager-fallback
max-ephones 24
max-dn 144
ip source-address 1.4.214.104 port 2000
voicemail 14011
call-forward busy 14011
call-forward noan 14011 timeout 3
!
! Cisco Unified SRST Voice-Mail Integration Pattern Configuration
!
vm-integration
pattern direct 2 CGN *
pattern ext-to-ext no-answer 5 FDN * CGN *
pattern ext-to-ext busy 7 FDN * CGN *
pattern trunk-to-ext no-answer 4 FDN * CGN *
pattern trunk-to-ext busy 6 FDN * CGN *
 

Configuring Voice-Mail Access over FXO and FXS: Example

The following example shows how to configure the Cisco Unified SCCP SRST router to forward unanswered calls to voice mail. In this example, the voice-mail number is 1101, the voice-mail system is connected to FXS voice port 1/1/1, and the voice mailbox numbers are 3001, 3002, and 3006.

voice-port 1/1/1
timing digit 250
timing inter-digit 250
 
dial-peer voice 1102 pots
destination-pattern 1101T
port 1/1/1
 
call-manager-fallback
timeouts interdigit 5
ip source-address 1.6.0.199 port 2000
max-ephones 24
max-dn 24
transfer-pattern 3...
voicemail 1101
call-forward busy 1101
call-forward noan 1101 timeout 3
moh minuet.au
 
vm-integration
pattern direct * CGN
pattern ext-to-ext no-answer # FDN #2
pattern ext-to-ext busy # FDN #2
pattern trunk-to-ext no-answer # FDN #2
pattern trunk-to-ext busy # FDN #2
 

Configuring Voice-Mail Access over BRI and PRI: Example

The following example shows how to configure the Cisco Unified SCCP SRST router to forward unanswered calls to voice mail. In this example, the voice-mail number is 1101, the voice-mail system is connected to a BRI or PRI voice port, and the voice mailbox numbers are 3001, 3002, and 3006.

controller T1 2/0
framing esf
clock source line primary
linecode b8zs
cablelength short 133
pri-group timeslots 21-24
 
interface Serial2/0:23
no ip address
no logging event link-status
isdn switch-type primary-net5
isdn incoming-voice voice
isdn T309-enable
no cdp enable
 
voice-port 2/0:23
 
dial-peer voice 1102 pots
destination-pattern 1101T
direct-inward-dial
port 2/0:23
 
call-manager-fallback
timeouts interdigit 5
ip source-address 1.6.0.199 port 2000
max-ephones 24
max-dn 24
transfer-pattern 3...
voicemail 1101
call-forward busy 1101
call-forward noan 1101 timeout 3
moh minuet.au

Message Waiting Indication for SIP SRST: Example

The following is an example of a NOTIFY message received at SRST indicating that there is a voice mail for extension 32002:

Received:
NOTIFY sip:32002@10.4.49.65:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.4.49.66:5060;branch=z9hG4bK.D6.7wAl9CN6khf305D1MQ~~194
Max-Forwards: 70
To: <sip:32002@10.4.49.65:5060>
From: <sip:32002@10.4.49.66:5060>;tag=dsd3d29b2f
Call-ID: f0e7ae97-1227@sip:32002@10.4.49.66:5060
CSeq: 1 NOTIFY
Content-Length: 112
Contact: <sip:32002@10.4.49.66:5060>
Content-Type: application/simple-message-summary
Event: message-summary
Messages-Waiting: yes
Message-Account: sip:32002@10.4.49.66
Voice-Message: 1/0 (1/0)
Fax-Message: 0/0 (0/0)

How to Configure DTMF Relay for SIP Applications and Voice Mail

DTMF relay for SIP applications can be used in two voice-mail situations:

For SIP SRST forwarding call to voicemail configuration, see the “” section.


Note Voice Mail number associate with SIP phone message button in SRST is configured by Cisco Unified Communications Manager (CUCM), and not configurable by SIP SRST. The administrator needs to know the voice mail number set by CUCM to configure proper dial peer to voice mail system in SIP SRST.


DTMF Relay Using SIP RFC 2833

Cisco Unified Skinny Client Control Protocol (SCCP) Phones, such as those used with Cisco Unified SRST systems, provide only out-of-band DTMF digit indications. To enable SCCP phones to send digit information to remote SIP-based IVR and voice-mail applications, Cisco Unified SRST 3.2 and later versions provide conversion from the out-of-band SCCP digit indication to the SIP standard for DTMF relay, which is RFC 2833. You select this method in the SIP VoIP dial peer using the dtmf-relay rtp-nte command.

The SIP DTMF relay method is needed in the following situations:

  • When SIP is used to connect a Cisco Unified SRST system to a remote SIP-based IVR or voice-mail application, such as Cisco Unity.
  • When SIP is used to connect a Cisco Unified SRST system to a remote SIP-PSTN voice gateway that goes through the PSTN to a voice-mail or IVR application.

NoteThe need to use out-of-band DTMF relay conversion is limited to SCCP phones. SIP phones natively support in-band DTMF relay as specified in RFC 2833. The need to use out-of-band DTMF relay conversion is limited to SCCP phones. SIP phones natively support in-band DTMF relay as specified in RFC 2833.


To enable SIP DTMF relay using RFC 2833, the commands in this section must be used on both originating and terminating gateways.

SUMMARY STEPS

1. dial-peer voice tag voip

2. dtmf-relay rtp-nte

3. exit

4. sip-ua

5. notify telephone-event max-duration time

6. exit

DETAILED STEPS

 

Command or Action
Purpose

Step 1

dial-peer voice tag voip

 

Router(config)# dial-peer voice 2 voip

Enters dial-peer configuration mode.

Step 2

dtmf-relay rtp-nte

 

Router(config-dial-peer)# dtmf-relay rtp-nte

Forwards DTMF tones by using Real-Time Transport Protocol (RTP) with the Named Telephone Event (NTE) payload type.

Step 3

exit

 

Router(config-dial-peer)# exit

Exits dial-peer configuration mode.

Step 4

sip-ua

 

Router(config)# sip-ua

Enables SIP user-agent configuration mode.

Step 5

notify telephone-event max-duration time

 

Router(config-sip-ua)# notify telephone-event max-duration 2000

Configures the maximum time interval allowed between two consecutive NOTIFY messages for a single DTMF event.

  • max-duration time : Time interval between consecutive NOTIFY messages for a single DTMF event, in milliseconds. Range is from 500 to 3000. Default is 2000.

Step 6

exit

 

Router(config-sip-ua)# exit

Exits SIP user-agent configuration mode.

Troubleshooting Tips

The dial-peer section of the show running-config command output displays DTMF relay status when it is configured, as shown in this excerpt:

dial-peer voice 123 voip
destination-pattern [12]...
monitor probe icmp-ping
session protocol sipv2
session target ipv4:10.8.17.42
dtmf-relay rtp-nte
 

DTMF Relay Using SIP Notify (Nonstandard)

To use voice mail on a SIP network that connects to a Cisco Unity Express system, use a nonstandard SIP Notify format. To configure the Notify format, use the sip-notify keyword with the dtmf-relay command. Using the sip-notify keyword may be required for backward compatibility with Cisco SRST Versions 3.0 and 3.1.

SUMMARY STEPS

1. dial-peer voice tag voip

2. dtmf-relay sip-notify

3. exit

4. sip-ua

5. notify telephone-event max-duration time

6. exit

DETAILED STEPS

 

Command or Action
Purpose

Step 1

dial-peer voice tag voip

 

Router(config)# dial-peer voice 2 voip

Enters dial-peer configuration mode.

Step 2

dtmf-relay sip-notify

 

Router(config-dial-peer)# dtmf-relay sip-notify

Forwards DTMF tones using SIP NOTIFY messages.

Step 3

exit

 

Router(config-dial-peer)# exit

Exits dial-peer configuration mode.

Step 4

sip-ua

 

Router(config)# sip-ua

Enables SIP user-agent configuration mode.

Step 5

notify telephone-event max-duration time

 

Router(config-sip-ua)# notify telephone-event max-duration 2000

Configures the maximum time interval allowed between two consecutive NOTIFY messages for a single DTMF event.

  • max-duration time : Time interval between consecutive NOTIFY messages for a single DTMF event, in milliseconds. Range is from 500 to 3000. Default is 2000.

Step 6

exit

 

Router(config-sip-ua)# exit

Exits SIP user-agent configuration mode.

Troubleshooting Tips

The show sip-ua status command output displays the time interval between consecutive NOTIFY messages for a telephone event. In the following example, the time interval is 2000 ms:

Router# show sip-ua status
SIP User Agent Status
SIP User Agent for UDP :ENABLED
SIP User Agent for TCP :ENABLED
SIP User Agent bind status(signaling):DISABLED
SIP User Agent bind status(media):DISABLED
SIP early-media for 180 responses with SDP:ENABLED
SIP max-forwards :6
SIP DNS SRV version:2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP:NONE
Check media source packets:DISABLED
Maximum duration for a telephone-event in NOTIFYs:2000 ms
SIP support for ISDN SUSPEND/RESUME:ENABLED
Redirection (3xx) message handling:ENABLED
 
SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported:audio image
Network types supported:IN
Address types supported:IP4
Transport types supported:RTP/AVP udptl

Where to Go Next

If you want to configure video parameters, see the “” section.

For additional information, see the “Additional References” section in the “l” chapter.